| Index: webrtc/audio/audio_send_stream.h
|
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
|
| index 5a2864b77fcbba116cd81691ab020c4f6a3e1252..93588d31d6f639fce4c978e676bb76d93e55d8dc 100644
|
| --- a/webrtc/audio/audio_send_stream.h
|
| +++ b/webrtc/audio/audio_send_stream.h
|
| @@ -43,7 +43,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
|
| // webrtc::AudioSendStream implementation.
|
| void Start() override;
|
| void Stop() override;
|
| - bool SendTelephoneEvent(int payload_type, int event,
|
| + bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
|
| int duration_ms) override;
|
| void SetMuted(bool muted) override;
|
| webrtc::AudioSendStream::Stats GetStats() const override;
|
|
|